The present invention relates to a packet transfer device and a communication system, and particularly to a packet transfer device (L2SW) and a communication system effective in reducing a transfer delay time when VOIP communication is performed through a virtual communication channel between SIP device terminals connected to the same packet transfer device in an Ethernet (registered trademark) network.
In recent years, there exists a case where an administrator of an apartment house such as a condominium, of a hotel, or of a multi-tenant building subscribes to a high-speed Internet connection service, and the administrator provides users with the shared Internet connection service which is used in common by the respective users through a LAN such as the Ethernet (registered trademark).
In the shared Internet connection service, it is unnecessary for each user to perform the Internet connection through an ISP and the user can easily use it, however, the connection is performed through the LAN such as the Ethernet (registered trademark), and a broad cast packet from a user terminal is transmitted to all user terminals through the L2SW. Thus, in the L2SW, the uplink VLAN setting of allowing packet transfer to, for example, a specific port (uplink port) connected to the ISP is performed for a port connected to each user terminal, so that the broad cast packet is not transferred to another user (port other than the uplink port), and the layer 2 communication between the ports connecting the users can be prevented. With respect to a packet other than the broad cast packet, the IP packet redirected from the uplink port by the uplink VLAN setting is routed back by an upper router, so that the communication between the users in the same domain is realized.
On the other hand, in the field of an IP telephone, there is known a Voice over IP (VOIP) technique in which voice is transmitted with IP packets. In the VOIP, a virtual communication path (session) is established between the terminals before the start of communication, and an IP packet including voice data is transferred on the communication path. The establishment and disconnection of the session between the terminals is performed in accordance with a session control protocol.
The Internet Engineering Task Force (IETF) specifies, as a session control protocol in IP multimedia communication, Session Initiation Protocol (SIP, see, for example, non-patent document 1: RFC3261) suitable for the VOIP and Real-time Transport Protocol (RTP, see, for example, non-patent document 3: RFC1889). The SIP is an application protocol using a transport mechanism such as Transmission Control Protocol (TCP) or User Datagram Protocol (UDP). Besides, the SIP is a text-base protocol, and a SIP message has a header part to transport request or response information, and a message body in which the session content is described. In the description of the session, for example, Session Description Protocol (SDP, see, for example, non-patent document 2: RFC2327) is applied, and a communication partner is identified by SIP Uniform Resource Identifier (URI). In the VOIP, the RTP is used for transferring encoded voice data.
Operation modes of a SIP server include a Proxy mode in which a session establishment (call setting) request between terminals is mediated by the SIP server, and a Redirect mode in which a sending terminal acquires information of a receiving terminal from the SIP server and directly communicates with the receiving terminal.